/*----------------------------------------------------------------------------
* Post - adaptive postfilter main function
*----------------------------------------------------------------------------
*/
void Post(
Word16 t0, /* input : pitch delay given by coder */
Word16 *signal_ptr, /* input : input signal (pointer to current subframe */
Word16 *coeff, /* input : LPC coefficients for current subframe */
Word16 *sig_out, /* output: postfiltered output */
Word16 *vo, /* output: voicing decision 0 = uv, > 0 delay */
Word16 Vad /* input : frame type */
)
{
/* Local variables and arrays */
Word16 apond1[MP1]; /* s.t. denominator coeff. */
Word16 sig_ltp[L_SUBFRP1]; /* H0 output signal */
Word16 *sig_ltp_ptr;
Word16 parcor0;
/* Compute weighted LPC coefficients */
Weight_Az(coeff, GAMMA1_PST, M, apond1);
Weight_Az(coeff, GAMMA2_PST, M, apond2);
/* Compute A(gamma2) residual */
Residu(apond2, signal_ptr, res2_ptr, L_SUBFR);
/* Harmonic filtering */
sig_ltp_ptr = sig_ltp + 1;
if (sub(Vad, 1) == 0)
pst_ltp(t0, res2_ptr, sig_ltp_ptr, vo);
else {
*vo = 0;
Copy(res2_ptr, sig_ltp_ptr, L_SUBFR);
}
/* Save last output of 1/A(gamma1) */
/* (from preceding subframe) */
sig_ltp[0] = *ptr_mem_stp;
/* Controls short term pst filter gain and compute parcor0 */
calc_st_filt(apond2, apond1, &parcor0, sig_ltp_ptr);
/* 1/A(gamma1) filtering, mem_stp is updated */
Syn_filt(apond1, sig_ltp_ptr, sig_ltp_ptr, L_SUBFR, mem_stp, 1);
/* Tilt filtering */
filt_mu(sig_ltp, sig_out, parcor0);
/* Gain control */
scale_st(signal_ptr, sig_out, &gain_prec);
/**** Update for next subframe */
Copy(&res2[L_SUBFR], &res2[0], MEM_RES2);
return;
}
void pre_big(
enum Mode mode, /* i : coder mode */
const Word16 gamma1[], /* i : spectral exp. factor 1 */
const Word16 gamma1_12k2[],/* i : spectral exp. factor 1 for EFR */
const Word16 gamma2[], /* i : spectral exp. factor 2 */
Word16 A_t[], /* i : A(z) unquantized, for 4 subframes, Q12 */
Word16 frameOffset, /* i : Start position in speech vector, Q0 */
Word16 speech[], /* i : speech, Q0 */
Word16 mem_w[], /* i/o: synthesis filter memory state, Q0 */
Word16 wsp[], /* o : weighted speech Q0 */
Flag *pOverflow /* o : overflow indicator */
)
{
Word16 Ap1[MP1]; /* A(z) with spectral expansion */
Word16 Ap2[MP1]; /* A(z) with spectral expansion */
const Word16 *g1; /* Pointer to correct gammma1 vector */
Word16 aOffset;
Word16 i;
if (mode <= MR795)
{
g1 = gamma1;
}
else
{
g1 = gamma1_12k2;
}
if (frameOffset > 0)
{
aOffset = 2 * MP1;
}
else
{
aOffset = 0;
}
/* process two subframes (which form the "big" subframe) */
for (i = 0; i < 2; i++)
{
Weight_Ai(&A_t[aOffset], g1, Ap1);
Weight_Ai(&A_t[aOffset], gamma2, Ap2);
Residu(Ap1, &speech[frameOffset], &wsp[frameOffset], L_SUBFR);
Syn_filt(Ap2, &wsp[frameOffset], &wsp[frameOffset], L_SUBFR, mem_w, 1);
aOffset = add(aOffset, MP1, pOverflow);
frameOffset = add(frameOffset, L_SUBFR, pOverflow);
}
return;
}
/*----------------------------------------------------------------------------
* calc_st_filt - computes impulse response of A(gamma2) / A(gamma1)
* controls gain : computation of energy impulse response as
* SUMn (abs (h[n])) and computes parcor0
*----------------------------------------------------------------------------
*/
static void calc_st_filt(
Word16 *apond2, /* input : coefficients of numerator */
Word16 *apond1, /* input : coefficients of denominator */
Word16 *parcor0, /* output: 1st parcor calcul. on composed filter */
Word16 *sig_ltp_ptr /* in/out: input of 1/A(gamma1) : scaled by 1/g0 */
)
{
Word16 h[LONG_H_ST];
Word32 L_temp, L_g0;
Word16 g0, temp;
int i;
/* compute i.r. of composed filter apond2 / apond1 */
Syn_filt(apond1, apond2, h, LONG_H_ST, mem_zero, 0);
/* compute 1st parcor */
calc_rc0_h(h, parcor0);
/* compute g0 */
L_g0 = 0L;
for(i=0; i<LONG_H_ST; i++) {
L_temp = L_deposit_l(abs_s(h[i]));
L_g0 = L_add(L_g0, L_temp);
}
g0 = extract_h(L_shl(L_g0, 14));
/* Scale signal input of 1/A(gamma1) */
if(sub(g0, 1024)>0) {
temp = div_s(1024, g0); /* temp = 2**15 / gain0 */
for(i=0; i<L_SUBFR; i++) {
sig_ltp_ptr[i] = mult_r(sig_ltp_ptr[i], temp);
}
}
return;
}
void Post_Filter(
Word16 *syn, /* in/out: synthesis speech (postfiltered is output) */
Word16 *Az_4, /* input : interpolated LPC parameters in all subframes */
Word16 *T, /* input : decoded pitch lags in all subframes */
Word16 Vad
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
Word16 res2_pst[L_SUBFR]; /* res2[] after pitch postfiltering */
Word16 syn_pst[L_FRAME]; /* post filtered synthesis speech */
Word16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
Word16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
Word16 t0_max, t0_min; /* closed-loop pitch search range */
Word16 i_subfr; /* index for beginning of subframe */
Word16 h[L_H];
Word16 i, j;
Word16 temp1, temp2;
Word32 L_tmp;
postfilt_type* ppost_filt = pg729dec->ppost_filt;
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find pitch range t0_min - t0_max */
t0_min = sub(*T++, 3);
t0_max = add(t0_min, 6);
if (sub(t0_max, PIT_MAX) > 0) {
t0_max = PIT_MAX;
t0_min = sub(t0_max, 6);
}
/* Find weighted filter coefficients Ap3[] and ap[4] */
Weight_Az(Az, GAMMA2_PST, M, Ap3);
Weight_Az(Az, GAMMA1_PST, M, Ap4);
/* filtering of synthesis speech by A(z/GAMMA2_PST) to find res2[] */
Residu(Ap3, &syn[i_subfr], ppost_filt->res2, L_SUBFR);
/* scaling of "res2[]" to avoid energy overflow */
for (j=0; j<L_SUBFR; j++)
{
ppost_filt->scal_res2[j] = shr(ppost_filt->res2[j], 2);
}
/* pitch postfiltering */
if (sub(Vad, 1) == 0)
pit_pst_filt(ppost_filt->res2, ppost_filt->scal_res2, t0_min, t0_max, L_SUBFR, res2_pst);
else
for (j=0; j<L_SUBFR; j++)
res2_pst[j] = ppost_filt->res2[j];
/* tilt compensation filter */
/* impulse response of A(z/GAMMA2_PST)/A(z/GAMMA1_PST) */
Copy(Ap3, h, M+1);
Set_zero(&h[M+1], L_H-M-1);
Syn_filt(Ap4, h, h, L_H, &h[M+1], 0);
/* 1st correlation of h[] */
L_tmp = L_mult(h[0], h[0]);
for (i=1; i<L_H; i++) L_tmp = L_mac(L_tmp, h[i], h[i]);
temp1 = extract_h(L_tmp);
L_tmp = L_mult(h[0], h[1]);
for (i=1; i<L_H-1; i++) L_tmp = L_mac(L_tmp, h[i], h[i+1]);
temp2 = extract_h(L_tmp);
if(temp2 <= 0) {
temp2 = 0;
}
else {
temp2 = mult(temp2, MU);
temp2 = div_s(temp2, temp1);
}
preemphasis(res2_pst, temp2, L_SUBFR);
/* filtering through 1/A(z/GAMMA1_PST) */
Syn_filt(Ap4, res2_pst, &syn_pst[i_subfr], L_SUBFR, ppost_filt->mem_syn_pst, 1);
/* scale output to input */
agc(&syn[i_subfr], &syn_pst[i_subfr], L_SUBFR);
/* update res2[] buffer; shift by L_SUBFR */
Copy(&ppost_filt->res2[L_SUBFR-PIT_MAX], &ppost_filt->res2[-PIT_MAX], PIT_MAX);
Copy(&ppost_filt->scal_res2[L_SUBFR-PIT_MAX], &ppost_filt->scal_res2[-PIT_MAX], PIT_MAX);
Az += MP1;
}
/* update syn[] buffer */
Copy(&syn[L_FRAME-M], &syn[-M], M);
/* overwrite synthesis speech by postfiltered synthesis speech */
Copy(syn_pst, syn, L_FRAME);
return;
}
void Coder_ld8a(
Word16 ana[], /* output : Analysis parameters */
Word16 frame, /* input : frame counter */
Word16 vad_enable /* input : VAD enable flag */
)
{
/* LPC analysis */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap_t[(MP1)*2]; /* A(z/gamma) for the 2 subframes */
Word16 *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* Other vectors */
Word16 h1[L_SUBFR]; /* Impulse response h1[] */
Word16 xn[L_SUBFR]; /* Target vector for pitch search */
Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
Word16 code[L_SUBFR]; /* Fixed codebook excitation */
Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */
Word16 g_coeff[4]; /* Correlations between xn & y1 */
Word16 g_coeff_cs[5];
Word16 exp_g_coeff_cs[5]; /* Correlations between xn, y1, & y2
<y1,y1>, -2<xn,y1>,
<y2,y2>, -2<xn,y2>, 2<y1,y2> */
/* Scalars */
Word16 i, j, k, i_subfr;
Word16 T_op, T0, T0_min, T0_max, T0_frac;
Word16 gain_pit, gain_code, index;
Word16 temp, taming;
Word32 L_temp;
/*------------------------------------------------------------------------*
* - Perform LPC analysis: *
* * autocorrelation + lag windowing *
* * Levinson-durbin algorithm to find a[] *
* * convert a[] to lsp[] *
* * quantize and code the LSPs *
* * find the interpolated LSPs and convert to a[] for the 2 *
* subframes (both quantized and unquantized) *
*------------------------------------------------------------------------*/
{
/* Temporary vectors */
Word16 r_l[NP+1], r_h[NP+1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
/* For G.729B */
Word16 rh_nbe[MP1];
Word16 lsf_new[M];
Word16 lsfq_mem[MA_NP][M];
Word16 exp_R0, Vad;
/* LP analysis */
Autocorr(p_window, NP, r_h, r_l, &exp_R0); /* Autocorrelations */
Copy(r_h, rh_nbe, MP1);
Lag_window(NP, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, Ap_t, rc, &temp); /* Levinson Durbin */
Az_lsp(Ap_t, lsp_new, lsp_old); /* From A(z) to lsp */
/* For G.729B */
/* ------ VAD ------- */
Lsp_lsf(lsp_new, lsf_new, M);
vad(rc[1], lsf_new, r_h, r_l, exp_R0, p_window, frame,
pastVad, ppastVad, &Vad);
Update_cng(rh_nbe, exp_R0, Vad);
/* ---------------------- */
/* Case of Inactive frame */
/* ---------------------- */
if ((Vad == 0) && (vad_enable == 1)){
Get_freq_prev(lsfq_mem);
Cod_cng(exc, pastVad, lsp_old_q, Aq_t, ana, lsfq_mem, &seed);
Update_freq_prev(lsfq_mem);
ppastVad = pastVad;
pastVad = Vad;
/* Update wsp, mem_w and mem_w0 */
Aq = Aq_t;
for(i_subfr=0; i_subfr < L_FRAME; i_subfr += L_SUBFR) {
/* Residual signal in xn */
Residu(Aq, &speech[i_subfr], xn, L_SUBFR);
Weight_Az(Aq, GAMMA1, M, Ap_t);
/* Compute wsp and mem_w */
Ap = Ap_t + MP1;
Ap[0] = 4096;
for(i=1; i<=M; i++) /* Ap[i] = Ap_t[i] - 0.7 * Ap_t[i-1]; */
Ap[i] = sub(Ap_t[i], mult(Ap_t[i-1], 22938));
Syn_filt(Ap, xn, &wsp[i_subfr], L_SUBFR, mem_w, 1);
/* Compute mem_w0 */
for(i=0; i<L_SUBFR; i++) {
xn[i] = sub(xn[i], exc[i_subfr+i]); /* residu[] - exc[] */
}
Syn_filt(Ap_t, xn, xn, L_SUBFR, mem_w0, 1);
Aq += MP1;
}
sharp = SHARPMIN;
/* Update memories for next frames */
Copy(&old_speech[L_FRAME], &old_speech[0], L_TOTAL-L_FRAME);
Copy(&old_wsp[L_FRAME], &old_wsp[0], PIT_MAX);
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
return;
} /* End of inactive frame case */
/* -------------------- */
/* Case of Active frame */
/* -------------------- */
/* Case of active frame */
*ana++ = 1;
seed = INIT_SEED;
ppastVad = pastVad;
pastVad = Vad;
/* LSP quantization */
Qua_lsp(lsp_new, lsp_new_q, ana);
ana += 2; /* Advance analysis parameters pointer */
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes *
* The interpolated parameters are in array Aq_t[]. *
*--------------------------------------------------------------------*/
Int_qlpc(lsp_old_q, lsp_new_q, Aq_t);
/* Compute A(z/gamma) */
Weight_Az(&Aq_t[0], GAMMA1, M, &Ap_t[0]);
Weight_Az(&Aq_t[MP1], GAMMA1, M, &Ap_t[MP1]);
/* update the LSPs for the next frame */
Copy(lsp_new, lsp_old, M);
Copy(lsp_new_q, lsp_old_q, M);
}
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
* - Find the open-loop pitch delay *
*----------------------------------------------------------------------*/
Residu(&Aq_t[0], &speech[0], &exc[0], L_SUBFR);
Residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);
{
Word16 Ap1[MP1];
Ap = Ap_t;
Ap1[0] = 4096;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
}
/* Find open loop pitch lag */
T_op = Pitch_ol_fast(wsp, PIT_MAX, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = sub(T_op, 3);
if (sub(T0_min,PIT_MIN)<0) {
T0_min = PIT_MIN;
}
T0_max = add(T0_min, 6);
if (sub(T0_max ,PIT_MAX)>0)
{
T0_max = PIT_MAX;
T0_min = sub(T0_max, 6);
}
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* To find the pitch and innovation parameters. The subframe size is *
* L_SUBFR and the loop is repeated 2 times. *
* - find the weighted LPC coefficients *
* - find the LPC residual signal res[] *
* - compute the target signal for pitch search *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - find the closed-loop pitch parameters *
* - encode the pitch delay *
* - find target vector for codebook search *
* - codebook search *
* - VQ of pitch and codebook gains *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
Ap = Ap_t; /* pointer to weighted LPC coefficients */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
h1[0] = 4096;
Set_zero(&h1[1], L_SUBFR-1);
Syn_filt(Ap, h1, h1, L_SUBFR, &h1[1], 0);
/*----------------------------------------------------------------------*
* Find the target vector for pitch search: *
*----------------------------------------------------------------------*/
Syn_filt(Ap, &exc[i_subfr], xn, L_SUBFR, mem_w0, 0);
/*---------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*---------------------------------------------------------------------*/
T0 = Pitch_fr3_fast(&exc[i_subfr], xn, h1, L_SUBFR, T0_min, T0_max,
i_subfr, &T0_frac);
index = Enc_lag3(T0, T0_frac, &T0_min, &T0_max,PIT_MIN,PIT_MAX,i_subfr);
*ana++ = index;
if (i_subfr == 0) {
*ana++ = Parity_Pitch(index);
}
/*-----------------------------------------------------------------*
* - find filtered pitch exc *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
Syn_filt(Ap, &exc[i_subfr], y1, L_SUBFR, mem_zero, 0);
gain_pit = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
taming = test_err(T0, T0_frac);
if( taming == 1){
if (sub(gain_pit, GPCLIP) > 0) {
gain_pit = GPCLIP;
}
}
/* xn2[i] = xn[i] - y1[i] * gain_pit */
for (i = 0; i < L_SUBFR; i++)
{
L_temp = L_mult(y1[i], gain_pit);
L_temp = L_shl(L_temp, 1); /* gain_pit in Q14 */
xn2[i] = sub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_Code_A(xn2, h1, T0, sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*-----------------------------------------------------*
* - Quantization of gains. *
*-----------------------------------------------------*/
g_coeff_cs[0] = g_coeff[0]; /* <y1,y1> */
exp_g_coeff_cs[0] = negate(g_coeff[1]); /* Q-Format:XXX -> JPN */
g_coeff_cs[1] = negate(g_coeff[2]); /* (xn,y1) -> -2<xn,y1> */
exp_g_coeff_cs[1] = negate(add(g_coeff[3], 1)); /* Q-Format:XXX -> JPN */
Corr_xy2( xn, y1, y2, g_coeff_cs, exp_g_coeff_cs ); /* Q0 Q0 Q12 ^Qx ^Q0 */
/* g_coeff_cs[3]:exp_g_coeff_cs[3] = <y2,y2> */
/* g_coeff_cs[4]:exp_g_coeff_cs[4] = -2<xn,y2> */
/* g_coeff_cs[5]:exp_g_coeff_cs[5] = 2<y1,y2> */
*ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
L_SUBFR, &gain_pit, &gain_code, taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
sharp = gain_pit;
if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters memories for finding the target *
* vector in the next subframe *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/* exc[i] in Q0 gain_pit in Q14 */
/* code[i] in Q13 gain_cod in Q1 */
L_temp = L_mult(exc[i+i_subfr], gain_pit);
L_temp = L_mac(L_temp, code[i], gain_code);
L_temp = L_shl(L_temp, 1);
exc[i+i_subfr] = round(L_temp);
}
update_exc_err(gain_pit, T0);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
temp = extract_h(L_shl( L_mult(y1[i], gain_pit), 1) );
k = extract_h(L_shl( L_mult(y2[i], gain_code), 2) );
mem_w0[j] = sub(xn[i], add(temp, k));
}
Aq += MP1; /* interpolated LPC parameters for next subframe */
Ap += MP1;
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME: *
* speech[], wsp[] and exc[] *
*--------------------------------------------------*/
Copy(&old_speech[L_FRAME], &old_speech[0], L_TOTAL-L_FRAME);
Copy(&old_wsp[L_FRAME], &old_wsp[0], PIT_MAX);
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
return;
}
void subframePostProc(
Word16 *speech, /* i : speech segment */
enum Mode mode, /* i : coder mode */
Word16 i_subfr, /* i : Subframe nr */
Word16 gain_pit, /* i : Pitch gain Q14 */
Word16 gain_code, /* i : Decoded innovation gain */
Word16 *Aq, /* i : A(z) quantized for the 4 subframes */
Word16 synth[], /* i : Local snthesis */
Word16 xn[], /* i : Target vector for pitch search */
Word16 code[], /* i : Fixed codebook exitation */
Word16 y1[], /* i : Filtered adaptive exitation */
Word16 y2[], /* i : Filtered fixed codebook excitation */
Word16 *mem_syn, /* i/o : memory of synthesis filter */
Word16 *mem_err, /* o : pointer to error signal */
Word16 *mem_w0, /* o : memory of weighting filter */
Word16 *exc, /* o : long term prediction residual */
Word16 *sharp, /* o : pitch sharpening value */
Flag *pOverflow /* o : overflow indicator */
)
{
Word16 i;
Word16 j;
Word16 temp;
Word32 L_temp;
Word32 L_temp2;
Word16 tempShift;
Word16 kShift;
Word16 pitch_fac;
Word16 *p_exc;
Word16 *p_code;
OSCL_UNUSED_ARG(pOverflow);
if (mode != MR122)
{
tempShift = 1;
kShift = 16 - 2 - 1;
pitch_fac = gain_pit;
}
else
{
tempShift = 2;
kShift = 16 - 4 - 1;
pitch_fac = gain_pit >> 1;
}
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
if (gain_pit < SHARPMAX)
{
*sharp = gain_pit;
}
else
{
*sharp = SHARPMAX;
}
/*------------------------------------------------------*
* - Find the total excitation *
* - find synthesis speech corresponding to exc[] *
* - update filters memories for finding the target *
* vector in the next subframe *
* (update error[-m..-1] and mem_w_err[]) *
*------------------------------------------------------*/
p_exc = &exc[ i_subfr];
p_code = &code[0];
for (i = L_SUBFR >> 1; i != 0 ; i--)
{
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/*
* 12k2 others
* ---------------------------------
* exc Q0 Q0
* gain_pit Q14 Q14
* pitch_fac Q13 Q14
* product: Q14 Q15
*
* code Q12 Q13
* gain_code Q1 Q1
* product Q14 Q15
* sum Q14 Q15
*
* tempShift 2 1
* sum<<tempShift Q16 Q16
* result -> exc Q0 Q0
*/
L_temp = ((Word32) * (p_exc++) * pitch_fac) << 1;
L_temp2 = ((Word32) * (p_exc--) * pitch_fac) << 1;
L_temp += ((Word32) * (p_code++) * gain_code) << 1;
L_temp2 += ((Word32) * (p_code++) * gain_code) << 1;
L_temp <<= tempShift;
L_temp2 <<= tempShift;
*(p_exc++) = (Word16)((L_temp + 0x08000L) >> 16);
*(p_exc++) = (Word16)((L_temp2 + 0x08000L) >> 16);
}
Syn_filt(
Aq,
&exc[i_subfr],
&synth[i_subfr],
L_SUBFR,
mem_syn,
1);
for (i = L_SUBFR - M, j = 0; i < L_SUBFR; i++, j++)
{
mem_err[j] = speech[i_subfr + i] - synth[i_subfr + i];
/*
* 12k2 others
* ---------------------------------
* y1 Q0 Q0
* gain_pit Q14 Q14
* product Q15 Q15
* shifted prod. Q16 Q16
* temp Q0 Q0
*
* y2 Q10 Q12
* gain_code Q1 Q1
* product Q12 Q14
* kshift 4 2
* shifted prod. Q16 Q16
* k Q0 Q0
* mem_w0,xn,sum Q0 Q0
*/
L_temp = ((Word32)y1[i] * gain_pit);
temp = (Word16)(L_temp >> 14);
L_temp = ((Word32)y2[i] * gain_code);
temp += (Word16)(L_temp >> kShift);
mem_w0[j] = xn[i] - temp;
}
return;
}
/*
**************************************************************************
* Function: Post_Filter
* Purpose: postfiltering of synthesis speech.
* Description:
* The postfiltering process is described as follows:
*
* - inverse filtering of syn[] through A(z/0.7) to get res2[]
* - tilt compensation filtering; 1 - MU*k*z^-1
* - synthesis filtering through 1/A(z/0.75)
* - adaptive gain control
*
**************************************************************************
*/
int Post_Filter (
Post_FilterState *st, /* i/o : post filter states */
enum Mode mode, /* i : AMR mode */
Word16 *syn, /* i/o : synthesis speech (postfiltered is output) */
Word16 *Az_4 /* i : interpolated LPC parameters in all subfr. */
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
Word16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
Word16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
Word16 i_subfr; /* index for beginning of subframe */
Word16 h[L_H];
Word16 i;
Word16 temp1, temp2;
Word32 L_tmp;
Word16 *syn_work = &st->synth_buf[M];
move16 ();
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Copy (syn, syn_work , L_FRAME);
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find weighted filter coefficients Ap3[] and ap[4] */
test ();
test ();
if (sub(mode, MR122) == 0 || sub(mode, MR102) == 0)
{
Weight_Ai (Az, gamma3_MR122, Ap3);
Weight_Ai (Az, gamma4_MR122, Ap4);
}
else
{
Weight_Ai (Az, gamma3, Ap3);
Weight_Ai (Az, gamma4, Ap4);
}
/* filtering of synthesis speech by A(z/0.7) to find res2[] */
Residu (Ap3, &syn_work[i_subfr], st->res2, L_SUBFR);
/* tilt compensation filter */
/* impulse response of A(z/0.7)/A(z/0.75) */
Copy (Ap3, h, M + 1);
Set_zero (&h[M + 1], L_H - M - 1);
Syn_filt (Ap4, h, h, L_H, &h[M + 1], 0);
/* 1st correlation of h[] */
L_tmp = L_mult (h[0], h[0]);
for (i = 1; i < L_H; i++)
{
L_tmp = L_mac (L_tmp, h[i], h[i]);
}
temp1 = extract_h (L_tmp);
L_tmp = L_mult (h[0], h[1]);
for (i = 1; i < L_H - 1; i++)
{
L_tmp = L_mac (L_tmp, h[i], h[i + 1]);
}
temp2 = extract_h (L_tmp);
test ();
if (temp2 <= 0)
{
temp2 = 0;
move16 ();
}
else
{
temp2 = mult (temp2, MU);
temp2 = div_s (temp2, temp1);
}
preemphasis (st->preemph_state, st->res2, temp2, L_SUBFR);
/* filtering through 1/A(z/0.75) */
Syn_filt (Ap4, st->res2, &syn[i_subfr], L_SUBFR, st->mem_syn_pst, 1);
/* scale output to input */
agc (st->agc_state, &syn_work[i_subfr], &syn[i_subfr],
AGC_FAC, L_SUBFR);
Az += MP1;
}
/* update syn_work[] buffer */
Copy (&syn_work[L_FRAME - M], &syn_work[-M], M);
return 0;
}
/*
********************************************************************************
* PUBLIC PROGRAM CODE
********************************************************************************
*/
int subframePostProc(
Word16 *speech, /* i : speech segment */
enum Mode mode, /* i : coder mode */
Word16 i_subfr, /* i : Subframe nr */
Word16 gain_pit, /* i : Pitch gain Q14 */
Word16 gain_code, /* i : Decoded innovation gain */
Word16 *Aq, /* i : A(z) quantized for the 4 subframes */
Word16 synth[], /* i : Local snthesis */
Word16 xn[], /* i : Target vector for pitch search */
Word16 code[], /* i : Fixed codebook exitation */
Word16 y1[], /* i : Filtered adaptive exitation */
Word16 y2[], /* i : Filtered fixed codebook excitation */
Word16 *mem_syn, /* i/o : memory of synthesis filter */
Word16 *mem_err, /* o : pointer to error signal */
Word16 *mem_w0, /* o : memory of weighting filter */
Word16 *exc, /* o : long term prediction residual */
Word16 *sharp /* o : pitch sharpening value */
)
{
Word16 i, j, k;
Word16 temp;
Word32 L_temp;
Word16 tempShift;
Word16 kShift;
Word16 pitch_fac;
if (sub(mode, MR122) != 0)
{
tempShift = 1;
kShift = 2;
pitch_fac = gain_pit;
}
else
{
tempShift = 2;
kShift = 4;
pitch_fac = shr (gain_pit, 1);
}
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
*sharp = gain_pit;
if (sub(*sharp, SHARPMAX) > 0)
{
*sharp = SHARPMAX;
}
/*------------------------------------------------------*
* - Find the total excitation *
* - find synthesis speech corresponding to exc[] *
* - update filters memories for finding the target *
* vector in the next subframe *
* (update error[-m..-1] and mem_w_err[]) *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++) {
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/*
* 12k2 others
* ---------------------------------
* exc Q0 Q0
* gain_pit Q14 Q14
* pitch_fac Q13 Q14
* product: Q14 Q15
*
* code Q12 Q13
* gain_code Q1 Q1
* product Q14 Q15
* sum Q14 Q15
*
* tempShift 2 1
* sum<<tempShift Q16 Q16
* result -> exc Q0 Q0
*/
L_temp = L_mult (exc[i + i_subfr], pitch_fac);
L_temp = L_mac (L_temp, code[i], gain_code);
L_temp = L_shl (L_temp, tempShift);
exc[i + i_subfr] = round (L_temp);
}
Syn_filt(Aq, &exc[i_subfr], &synth[i_subfr], L_SUBFR,
mem_syn, 1);
for (i = L_SUBFR - M, j = 0; i < L_SUBFR; i++, j++) {
mem_err[j] = sub(speech[i_subfr + i],
synth[i_subfr + i]);
/*
* 12k2 others
* ---------------------------------
* y1 Q0 Q0
* gain_pit Q14 Q14
* product Q15 Q15
* shifted prod. Q16 Q16
* temp Q0 Q0
*
* y2 Q10 Q12
* gain_code Q1 Q1
* product Q12 Q14
* kshift 4 2
* shifted prod. Q16 Q16
* k Q0 Q0
* mem_w0,xn,sum Q0 Q0
*/
temp = extract_h(L_shl(L_mult(y1[i], gain_pit), 1));
k = extract_h(L_shl(L_mult(y2[i], gain_code), kShift));
mem_w0[j] = sub(xn[i], add(temp, k));
}
return 0;
}
void Coder_ld8a(
g729a_encoder_state *state,
Word16 ana[] /* output : Analysis parameters */
)
{
/* LPC analysis */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap_t[(MP1)*2]; /* A(z/gamma) for the 2 subframes */
Word16 *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* Other vectors */
Word16 h1[L_SUBFR]; /* Impulse response h1[] */
Word16 xn[L_SUBFR]; /* Target vector for pitch search */
Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
Word16 code[L_SUBFR]; /* Fixed codebook excitation */
Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */
Word16 g_coeff[4]; /* Correlations between xn & y1 */
Word16 g_coeff_cs[5];
Word16 exp_g_coeff_cs[5]; /* Correlations between xn, y1, & y2
<y1,y1>, -2<xn,y1>,
<y2,y2>, -2<xn,y2>, 2<y1,y2> */
/* Scalars */
Word16 i, j, k, i_subfr;
Word16 T_op, T0, T0_min, T0_max, T0_frac;
Word16 gain_pit, gain_code, index;
Word16 temp, taming;
Word32 L_temp;
/*------------------------------------------------------------------------*
* - Perform LPC analysis: *
* * autocorrelation + lag windowing *
* * Levinson-durbin algorithm to find a[] *
* * convert a[] to lsp[] *
* * quantize and code the LSPs *
* * find the interpolated LSPs and convert to a[] for the 2 *
* subframes (both quantized and unquantized) *
*------------------------------------------------------------------------*/
{
/* Temporary vectors */
Word16 r_l[MP1], r_h[MP1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
/* LP analysis */
Autocorr(state->p_window, M, r_h, r_l); /* Autocorrelations */
Lag_window(M, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, Ap_t, rc); /* Levinson Durbin */
Az_lsp(Ap_t, lsp_new, state->lsp_old); /* From A(z) to lsp */
/* LSP quantization */
Qua_lsp(state, lsp_new, lsp_new_q, ana);
ana += 2; /* Advance analysis parameters pointer */
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes *
* The interpolated parameters are in array Aq_t[]. *
*--------------------------------------------------------------------*/
Int_qlpc(state->lsp_old_q, lsp_new_q, Aq_t);
/* Compute A(z/gamma) */
Weight_Az(&Aq_t[0], GAMMA1, M, &Ap_t[0]);
Weight_Az(&Aq_t[MP1], GAMMA1, M, &Ap_t[MP1]);
/* update the LSPs for the next frame */
Copy(lsp_new, state->lsp_old, M);
Copy(lsp_new_q, state->lsp_old_q, M);
}
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
* - Find the open-loop pitch delay *
*----------------------------------------------------------------------*/
Residu(&Aq_t[0], &(state->speech[0]), &(state->exc[0]), L_SUBFR);
Residu(&Aq_t[MP1], &(state->speech[L_SUBFR]), &(state->exc[L_SUBFR]), L_SUBFR);
{
Word16 Ap1[MP1];
Ap = Ap_t;
Ap1[0] = 4096;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &(state->exc[0]), &(state->wsp[0]), L_SUBFR, state->mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &(state->exc[L_SUBFR]), &(state->wsp[L_SUBFR]), L_SUBFR, state->mem_w, 1);
}
/* Find open loop pitch lag */
T_op = Pitch_ol_fast(state->wsp, PIT_MAX, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = T_op - 3;
T0_max = T0_min + 6;
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
T0_max = PIT_MIN + 6;
}
else if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = PIT_MAX - 6;
}
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* To find the pitch and innovation parameters. The subframe size is *
* L_SUBFR and the loop is repeated 2 times. *
* - find the weighted LPC coefficients *
* - find the LPC residual signal res[] *
* - compute the target signal for pitch search *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - find the closed-loop pitch parameters *
* - encode the pitch delay *
* - find target vector for codebook search *
* - codebook search *
* - VQ of pitch and codebook gains *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
Ap = Ap_t; /* pointer to weighted LPC coefficients */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
h1[0] = 4096;
Set_zero(&h1[1], L_SUBFR-1);
Syn_filt(Ap, h1, h1, L_SUBFR, &h1[1], 0);
/*----------------------------------------------------------------------*
* Find the target vector for pitch search: *
*----------------------------------------------------------------------*/
Syn_filt(Ap, &(state->exc[i_subfr]), xn, L_SUBFR, state->mem_w0, 0);
/*---------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*---------------------------------------------------------------------*/
T0 = Pitch_fr3_fast(&(state->exc[i_subfr]), xn, h1, L_SUBFR, T0_min, T0_max,
i_subfr, &T0_frac);
index = Enc_lag3(T0, T0_frac, &T0_min, &T0_max,PIT_MIN,PIT_MAX,i_subfr);
*ana++ = index;
if (i_subfr == 0) {
*ana++ = Parity_Pitch(index);
}
/*-----------------------------------------------------------------*
* - find filtered pitch exc *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
Syn_filt(Ap, &(state->exc[i_subfr]), y1, L_SUBFR, state->mem_zero, 0);
gain_pit = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
taming = test_err(state, T0, T0_frac);
if( taming == 1){
if (gain_pit > GPCLIP) {
gain_pit = GPCLIP;
}
}
/* xn2[i] = xn[i] - y1[i] * gain_pit */
for (i = 0; i < L_SUBFR; i++)
{
//L_temp = L_mult(y1[i], gain_pit);
//L_temp = L_shl(L_temp, 1); /* gain_pit in Q14 */
L_temp = ((Word32)y1[i] * gain_pit) << 2;
xn2[i] = sub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_Code_A(xn2, h1, T0, state->sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*-----------------------------------------------------*
* - Quantization of gains. *
*-----------------------------------------------------*/
g_coeff_cs[0] = g_coeff[0]; /* <y1,y1> */
exp_g_coeff_cs[0] = negate(g_coeff[1]); /* Q-Format:XXX -> JPN */
g_coeff_cs[1] = negate(g_coeff[2]); /* (xn,y1) -> -2<xn,y1> */
exp_g_coeff_cs[1] = negate(add(g_coeff[3], 1)); /* Q-Format:XXX -> JPN */
Corr_xy2( xn, y1, y2, g_coeff_cs, exp_g_coeff_cs ); /* Q0 Q0 Q12 ^Qx ^Q0 */
/* g_coeff_cs[3]:exp_g_coeff_cs[3] = <y2,y2> */
/* g_coeff_cs[4]:exp_g_coeff_cs[4] = -2<xn,y2> */
/* g_coeff_cs[5]:exp_g_coeff_cs[5] = 2<y1,y2> */
*ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
L_SUBFR, &gain_pit, &gain_code, taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
state->sharp = gain_pit;
if (state->sharp > SHARPMAX) { state->sharp = SHARPMAX; }
else if (state->sharp < SHARPMIN) { state->sharp = SHARPMIN; }
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters memories for finding the target *
* vector in the next subframe *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/* exc[i] in Q0 gain_pit in Q14 */
/* code[i] in Q13 gain_cod in Q1 */
//L_temp = L_mult(exc[i+i_subfr], gain_pit);
//L_temp = L_mac(L_temp, code[i], gain_code);
//L_temp = L_shl(L_temp, 1);
L_temp = (Word32)(state->exc[i+i_subfr]) * (Word32)gain_pit +
(Word32)code[i] * (Word32)gain_code;
L_temp <<= 2;
state->exc[i+i_subfr] = g_round(L_temp);
}
update_exc_err(state, gain_pit, T0);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
temp = ((Word32)y1[i] * (Word32)gain_pit) >> 14;
k = ((Word32)y2[i] * (Word32)gain_code) >> 13;
state->mem_w0[j] = sub(xn[i], add(temp, k));
}
Aq += MP1; /* interpolated LPC parameters for next subframe */
Ap += MP1;
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME: *
* speech[], wsp[] and exc[] *
*--------------------------------------------------*/
Copy(&(state->old_speech[L_FRAME]), &(state->old_speech[0]), L_TOTAL-L_FRAME);
Copy(&(state->old_wsp[L_FRAME]), &(state->old_wsp[0]), PIT_MAX);
Copy(&(state->old_exc[L_FRAME]), &(state->old_exc[0]), PIT_MAX+L_INTERPOL);
}
/*
********************************************************************************
* PUBLIC PROGRAM CODE
********************************************************************************
*/
int subframePreProc(
enum Mode mode, /* i : coder mode */
const Word16 gamma1[], /* i : spectral exp. factor 1 */
const Word16 gamma1_12k2[],/* i : spectral exp. factor 1 for EFR */
const Word16 gamma2[], /* i : spectral exp. factor 2 */
Word16 *A, /* i : A(z) unquantized for the 4 subframes */
Word16 *Aq, /* i : A(z) quantized for the 4 subframes */
Word16 *speech, /* i : speech segment */
Word16 *mem_err, /* i : pointer to error signal */
Word16 *mem_w0, /* i : memory of weighting filter */
Word16 *zero, /* i : pointer to zero vector */
Word16 ai_zero[], /* o : history of weighted synth. filter */
Word16 exc[], /* o : long term prediction residual */
Word16 h1[], /* o : impulse response */
Word16 xn[], /* o : target vector for pitch search */
Word16 res2[], /* o : long term prediction residual */
Word16 error[] /* o : error of LPC synthesis filter */
)
{
Word16 i;
Word16 Ap1[MP1]; /* A(z) with spectral expansion */
Word16 Ap2[MP1]; /* A(z) with spectral expansion */
const Word16 *g1; /* Pointer to correct gammma1 vector */
/*---------------------------------------------------------------*
* mode specific pointer to gamma1 values *
*---------------------------------------------------------------*/
test (); test ();
if ( sub(mode, MR122) == 0 || sub(mode, MR102) == 0 )
{
g1 = gamma1_12k2; move16 ();
}
else
{
g1 = gamma1; move16 ();
}
/*---------------------------------------------------------------*
* Find the weighted LPC coefficients for the weighting filter. *
*---------------------------------------------------------------*/
Weight_Ai(A, g1, Ap1);
Weight_Ai(A, gamma2, Ap2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
for (i = 0; i <= M; i++)
{
ai_zero[i] = Ap1[i]; move16 ();
}
Syn_filt(Aq, ai_zero, h1, L_SUBFR, zero, 0);
Syn_filt(Ap2, h1, h1, L_SUBFR, zero, 0);
/*------------------------------------------------------------------------*
* *
* Find the target vector for pitch search: *
* *
*------------------------------------------------------------------------*/
/* LPC residual */
Residu(Aq, speech, res2, L_SUBFR);
Copy(res2, exc, L_SUBFR);
Syn_filt(Aq, exc, error, L_SUBFR, mem_err, 0);
Residu(Ap1, error, xn, L_SUBFR);
/* target signal xn[]*/
Syn_filt(Ap2, xn, xn, L_SUBFR, mem_w0, 0);
return 0;
}
void Post_Filter (
INT16 *syn, /* in/out: synthesis speech (postfiltered is output) */
INT16 *Az_4 /* input: interpolated LPC parameters in all subframes */
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
INT16 syn_pst[L_FRAME]; /* post filtered synthesis speech */
INT16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
INT16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
INT16 i_subfr; /* index for beginning of subframe */
INT16 h[L_H];
INT16 i;
INT16 temp1, temp2;
//INT32 L_tmp;
register INT32 tmp_hi=0;
register UINT32 tmp_lo=0;
VPP_EFR_PROFILE_FUNCTION_ENTER(Post_Filter);
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find weighted filter coefficients Ap3[] and ap[4] */
Weight_Ai (Az, F_gamma3, Ap3);
Weight_Ai (Az, F_gamma4, Ap4);
/* filtering of synthesis speech by A(z/0.7) to find res2[] */
Residu (Ap3, &syn[i_subfr], res2, L_SUBFR);
/* tilt compensation filter */
/* impulse response of A(z/0.7)/A(z/0.75) */
Copy (Ap3, h, M + 1);
//Set_zero (&h[M + 1], L_H - M - 1);
memset ((INT8*)&h[M + 1], 0, (L_H - M - 1)<<1);
Syn_filt (Ap4, h, h, L_H, &h[M + 1], 0);
/* 1st correlation of h[] */
//L_tmp = L_MULT(h[0], h[0]);
VPP_MLX16(tmp_hi,tmp_lo, h[0], h[0]);
for (i = 1; i < L_H; i++)
{
//L_tmp = L_MAC(L_tmp, h[i], h[i]);
VPP_MLA16(tmp_hi,tmp_lo, h[i], h[i]);
}
//temp1 = extract_h (L_tmp);
//temp1 = EXTRACT_H(VPP_SCALE64_TO_16(tmp_hi,tmp_lo));
temp1 = L_SHR_D((INT32)tmp_lo, 15);
//L_tmp = L_MULT(h[0], h[1]);
VPP_MLX16(tmp_hi,tmp_lo, h[0], h[1]);
for (i = 1; i < L_H - 1; i++)
{
//L_tmp = L_MAC(L_tmp, h[i], h[i + 1]);
VPP_MLA16(tmp_hi,tmp_lo, h[i], h[i + 1]);
}
//temp2 = extract_h (L_tmp);
//temp2 = EXTRACT_H(VPP_SCALE64_TO_16(tmp_hi,tmp_lo));
temp2 = L_SHR_D((INT32)tmp_lo, 15);
if (temp2 <= 0)
{
temp2 = 0;
}
else
{
//temp2 = mult (temp2, MU);
temp2 = MULT(temp2, MU);
temp2 = div_s (temp2, temp1);
}
preemphasis (res2, temp2, L_SUBFR);
/* filtering through 1/A(z/0.75) */
Syn_filt (Ap4, res2, &syn_pst[i_subfr], L_SUBFR, mem_syn_pst, 1);
/* scale output to input */
agc (&syn[i_subfr], &syn_pst[i_subfr], AGC_FAC, L_SUBFR);
Az += MP1;
}
/* update syn[] buffer */
Copy (&syn[L_FRAME - M], &syn[-M], M);
/* overwrite synthesis speech by postfiltered synthesis speech */
Copy (syn_pst, syn, L_FRAME);
VPP_EFR_PROFILE_FUNCTION_EXIT(Post_Filter);
return;
}
_declspec(dllexport) void Decod_ld8k(
Word16 parm[], /* (i) : vector of synthesis parameters
parm[0] = bad frame indicator (bfi) */
Word16 voicing, /* (i) : voicing decision from previous frame */
Word16 synth[], /* (o) : synthesis speech */
Word16 A_t[], /* (o) : decoded LP filter in 2 subframes */
Word16 *T0_first, /* (o) : decoded pitch lag in first subframe */
Word16 *Vad /* (o) : frame type */
)
{
Word16 *Az; /* Pointer on A_t */
Word16 lsp_new[M]; /* LSPs */
Word16 code[L_SUBFR]; /* ACELP codevector */
/* Scalars */
Word16 i, j, i_subfr;
Word16 T0, T0_frac, index;
Word16 bfi;
Word32 L_temp;
Word16 g_p, g_c; /* fixed and adaptive codebook gain */
Word16 bad_pitch; /* bad pitch indicator */
extern Flag g729Overflow;
/* for G.729B */
Word16 ftyp;
Word16 lsfq_mem[MA_NP][M];
/* Test bad frame indicator (bfi) */
bfi = *parm++;
/* for G.729B */
ftyp = *parm;
if(bfi == 1) {
if(past_ftyp == 1) ftyp = 1;
else ftyp = 0;
*parm = ftyp; /* modification introduced in version V1.3 */
}
*Vad = ftyp;
/* Processing non active frames (SID & not transmitted) */
if(ftyp != 1) {
Get_decfreq_prev(lsfq_mem);
Dec_cng(past_ftyp, sid_sav, sh_sid_sav, parm, exc, lsp_old,
A_t, &seed, lsfq_mem);
Update_decfreq_prev(lsfq_mem);
Az = A_t;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR) {
g729Overflow = 0;
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0);
if(g729Overflow != 0) {
/* In case of overflow in the synthesis */
/* -> Scale down vector exc[] and redo synthesis */
for(i=0; i<PIT_MAX+L_INTERPOL+L_FRAME; i++)
old_exc[i] = g729shr(old_exc[i], 2);
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);
}
else
Copy(&synth[i_subfr+L_SUBFR-M], mem_syn, M);
Az += MP1;
*T0_first = old_T0;
}
sharp = SHARPMIN;
}
/* Processing active frame */
else {
seed = INIT_SEED;
parm++;
/* Decode the LSPs */
D_lsp(parm, lsp_new, bfi);
parm += 2;
/* Interpolation of LPC for the 2 subframes */
Int_qlpc(lsp_old, lsp_new, A_t);
/* update the LSFs for the next frame */
Copy(lsp_new, lsp_old, M);
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR *
* times *
* - decode the pitch delay *
* - decode algebraic code *
* - decode pitch and codebook gains *
* - find the excitation and compute synthesis speech *
*------------------------------------------------------------------------*/
Az = A_t; /* pointer to interpolated LPC parameters */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
index = *parm++; /* pitch index */
if(i_subfr == 0)
{
i = *parm++; /* get parity check result */
bad_pitch = g729add(bfi, i);
if( bad_pitch == 0)
{
Dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else /* Bad frame, or parity error */
{
T0 = old_T0;
T0_frac = 0;
old_T0 = g729add( old_T0, 1);
if( g729sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
*T0_first = T0; /* If first frame */
}
else /* second subframe */
{
if( bfi == 0)
{
Dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else
{
T0 = old_T0;
T0_frac = 0;
old_T0 = g729add( old_T0, 1);
if( g729sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
}
/*-------------------------------------------------*
* - Find the adaptive codebook vector. *
*-------------------------------------------------*/
Pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);
/*-------------------------------------------------------*
* - Decode innovative codebook. *
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
if(bfi != 0) /* Bad frame */
{
parm[0] = Random(&seed_fer) & (Word16)0x1fff; /* 13 bits random */
parm[1] = Random(&seed_fer) & (Word16)0x000f; /* 4 bits random */
}
Decod_ACELP(parm[1], parm[0], code);
parm +=2;
j = g729shl(sharp, 1); /* From Q14 to Q15 */
if(g729sub(T0, L_SUBFR) <0 ) {
for (i = T0; i < L_SUBFR; i++) {
code[i] = g729add(code[i], g729mult(code[i-T0], j));
}
}
/*-------------------------------------------------*
* - Decode pitch and codebook gains. *
*-------------------------------------------------*/
index = *parm++; /* index of energy VQ */
Dec_gain(index, code, L_SUBFR, bfi, &gain_pitch, &gain_code);
/*-------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pitch *
*-------------------------------------------------------------*/
sharp = gain_pitch;
if (g729sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (g729sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*-------------------------------------------------------*
* - Find the total excitation. *
* - Find synthesis speech corresponding to exc[]. *
*-------------------------------------------------------*/
if(bfi != 0) /* Bad frame */
{
if (voicing == 0 ) {
g_p = 0;
g_c = gain_code;
} else {
g_p = gain_pitch;
g_c = 0;
}
} else {
g_p = gain_pitch;
g_c = gain_code;
}
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = g_p*exc[i] + g_c*code[i]; */
/* exc[i] in Q0 g_p in Q14 */
/* code[i] in Q13 g_code in Q1 */
L_temp = g729L_mult(exc[i+i_subfr], g_p);
L_temp = g729L_mac(L_temp, code[i], g_c);
L_temp = g729L_shl(L_temp, 1);
exc[i+i_subfr] = g729round(L_temp);
}
g729Overflow = 0;
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0);
if(g729Overflow != 0)
{
/* In case of overflow in the synthesis */
/* -> Scale down vector exc[] and redo synthesis */
for(i=0; i<PIT_MAX+L_INTERPOL+L_FRAME; i++)
old_exc[i] = g729shr(old_exc[i], 2);
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);
}
else
Copy(&synth[i_subfr+L_SUBFR-M], mem_syn, M);
Az += MP1; /* interpolated LPC parameters for next subframe */
}
}
/*------------*
* For G729b
*-----------*/
if(bfi == 0) {
L_temp = 0L;
for(i=0; i<L_FRAME; i++) {
L_temp = g729L_mac(L_temp, exc[i], exc[i]);
} /* may overflow => last level of SID quantizer */
sh_sid_sav = g729norm_l(L_temp);
sid_sav = g729round(g729L_shl(L_temp, sh_sid_sav));
sh_sid_sav = g729sub(16, sh_sid_sav);
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME exc[] *
*--------------------------------------------------*/
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
/* for G729b */
past_ftyp = ftyp;
return;
}
/*
**************************************************************************
*
* Function : dtx_dec
*
**************************************************************************
*/
int dtx_dec(
dtx_decState *st, /* i/o : State struct */
Word16 mem_syn[], /* i/o : AMR decoder state */
D_plsfState* lsfState, /* i/o : decoder lsf states */
gc_predState* predState, /* i/o : prediction states */
Cb_gain_averageState* averState, /* i/o : CB gain average states */
enum DTXStateType new_state, /* i : new DTX state */
enum Mode mode, /* i : AMR mode */
Word16 parm[], /* i : Vector of synthesis parameters */
Word16 synth[], /* o : synthesised speech */
Word16 A_t[] /* o : decoded LP filter in 4 subframes*/
)
{
Word16 log_en_index;
Word16 i, j;
Word16 int_fac;
Word32 L_log_en_int;
Word16 lsp_int[M];
Word16 log_en_int_e;
Word16 log_en_int_m;
Word16 level;
Word16 acoeff[M + 1];
Word16 refl[M];
Word16 pred_err;
Word16 ex[L_SUBFR];
Word16 ma_pred_init;
Word16 log_pg_e, log_pg_m;
Word16 log_pg;
Flag negative;
Word16 lsf_mean;
Word32 L_lsf_mean;
Word16 lsf_variab_index;
Word16 lsf_variab_factor;
Word16 lsf_int[M];
Word16 lsf_int_variab[M];
Word16 lsp_int_variab[M];
Word16 acoeff_variab[M + 1];
Word16 lsf[M];
Word32 L_lsf[M];
Word16 ptr;
Word16 tmp_int_length;
/* This function is called if synthesis state is not SPEECH
* the globally passed inputs to this function are
* st->sid_frame
* st->valid_data
* st->dtxHangoverAdded
* new_state (SPEECH, DTX, DTX_MUTE)
*/
test(); test();
if ((st->dtxHangoverAdded != 0) &&
(st->sid_frame != 0))
{
/* sid_first after dtx hangover period */
/* or sid_upd after dtxhangover */
/* set log_en_adjust to correct value */
st->log_en_adjust = dtx_log_en_adjust[mode];
ptr = add(st->lsf_hist_ptr, M); move16();
test();
if (sub(ptr, 80) == 0)
{
ptr = 0; move16();
}
Copy( &st->lsf_hist[st->lsf_hist_ptr],&st->lsf_hist[ptr],M);
ptr = add(st->log_en_hist_ptr,1); move16();
test();
if (sub(ptr, DTX_HIST_SIZE) == 0)
{
ptr = 0; move16();
}
move16();
st->log_en_hist[ptr] = st->log_en_hist[st->log_en_hist_ptr]; /* Q11 */
/* compute mean log energy and lsp *
* from decoded signal (SID_FIRST) */
st->log_en = 0; move16();
for (i = 0; i < M; i++)
{
L_lsf[i] = 0; move16();
}
/* average energy and lsp */
for (i = 0; i < DTX_HIST_SIZE; i++)
{
st->log_en = add(st->log_en,
shr(st->log_en_hist[i],3));
for (j = 0; j < M; j++)
{
L_lsf[j] = L_add(L_lsf[j],
L_deposit_l(st->lsf_hist[i * M + j]));
}
}
for (j = 0; j < M; j++)
{
lsf[j] = extract_l(L_shr(L_lsf[j],3)); /* divide by 8 */ move16();
}
Lsf_lsp(lsf, st->lsp, M);
/* make log_en speech coder mode independent */
/* added again later before synthesis */
st->log_en = sub(st->log_en, st->log_en_adjust);
/* compute lsf variability vector */
Copy(st->lsf_hist, st->lsf_hist_mean, 80);
for (i = 0; i < M; i++)
{
L_lsf_mean = 0; move32();
/* compute mean lsf */
for (j = 0; j < 8; j++)
{
L_lsf_mean = L_add(L_lsf_mean,
L_deposit_l(st->lsf_hist_mean[i+j*M]));
}
lsf_mean = extract_l(L_shr(L_lsf_mean, 3)); move16();
/* subtract mean and limit to within reasonable limits *
* moreover the upper lsf's are attenuated */
for (j = 0; j < 8; j++)
{
/* subtract mean */
st->lsf_hist_mean[i+j*M] =
sub(st->lsf_hist_mean[i+j*M], lsf_mean);
/* attenuate deviation from mean, especially for upper lsf's */
st->lsf_hist_mean[i+j*M] =
mult(st->lsf_hist_mean[i+j*M], lsf_hist_mean_scale[i]);
/* limit the deviation */
test();
if (st->lsf_hist_mean[i+j*M] < 0)
{
negative = 1; move16();
}
else
{
negative = 0; move16();
}
st->lsf_hist_mean[i+j*M] = abs_s(st->lsf_hist_mean[i+j*M]);
/* apply soft limit */
test();
if (sub(st->lsf_hist_mean[i+j*M], 655) > 0)
{
st->lsf_hist_mean[i+j*M] =
add(655, shr(sub(st->lsf_hist_mean[i+j*M], 655), 2));
}
/* apply hard limit */
test();
if (sub(st->lsf_hist_mean[i+j*M], 1310) > 0)
{
st->lsf_hist_mean[i+j*M] = 1310; move16();
}
test();
if (negative != 0)
{
st->lsf_hist_mean[i+j*M] = -st->lsf_hist_mean[i+j*M];move16();
}
}
}
}
test();
if (st->sid_frame != 0 )
{
/* Set old SID parameters, always shift */
/* even if there is no new valid_data */
Copy(st->lsp, st->lsp_old, M);
st->old_log_en = st->log_en; move16();
test();
if (st->valid_data != 0 ) /* new data available (no CRC) */
{
/* Compute interpolation factor, since the division only works *
* for values of since_last_sid < 32 we have to limit the *
* interpolation to 32 frames */
tmp_int_length = st->since_last_sid; move16();
st->since_last_sid = 0; move16();
test();
if (sub(tmp_int_length, 32) > 0)
{
tmp_int_length = 32; move16();
}
test();
if (sub(tmp_int_length, 2) >= 0)
{
move16();
st->true_sid_period_inv = div_s(1 << 10, shl(tmp_int_length, 10));
}
else
{
st->true_sid_period_inv = 1 << 14; /* 0.5 it Q15 */ move16();
}
Init_D_plsf_3(lsfState, parm[0]); /* temporay initialization */
D_plsf_3(lsfState, MRDTX, 0, &parm[1], st->lsp);
Set_zero(lsfState->past_r_q, M); /* reset for next speech frame */
log_en_index = parm[4]; move16();
/* Q11 and divide by 4 */
st->log_en = shl(log_en_index, (11 - 2)); move16();
/* Subtract 2.5 in Q11 */
st->log_en = sub(st->log_en, (2560 * 2));
/* Index 0 is reserved for silence */
test();
if (log_en_index == 0)
{
st->log_en = MIN_16; move16();
}
/* no interpolation at startup after coder reset */
/* or when SID_UPD has been received right after SPEECH */
test(); test();
if ((st->data_updated == 0) ||
(sub(st->dtxGlobalState, SPEECH) == 0)
)
{
Copy(st->lsp, st->lsp_old, M);
st->old_log_en = st->log_en; move16();
}
} /* endif valid_data */
/* initialize gain predictor memory of other modes */
ma_pred_init = sub(shr(st->log_en,1), 9000); move16();
test();
if (ma_pred_init > 0)
{
ma_pred_init = 0; move16();
}
test();
if (sub(ma_pred_init, -14436) < 0)
{
ma_pred_init = -14436; move16();
}
predState->past_qua_en[0] = ma_pred_init; move16();
predState->past_qua_en[1] = ma_pred_init; move16();
predState->past_qua_en[2] = ma_pred_init; move16();
predState->past_qua_en[3] = ma_pred_init; move16();
/* past_qua_en for other modes than MR122 */
ma_pred_init = mult(5443, ma_pred_init);
/* scale down by factor 20*log10(2) in Q15 */
predState->past_qua_en_MR122[0] = ma_pred_init; move16();
predState->past_qua_en_MR122[1] = ma_pred_init; move16();
predState->past_qua_en_MR122[2] = ma_pred_init; move16();
predState->past_qua_en_MR122[3] = ma_pred_init; move16();
} /* endif sid_frame */
/* CN generation */
/* recompute level adjustment factor Q11 *
* st->log_en_adjust = 0.9*st->log_en_adjust + *
* 0.1*dtx_log_en_adjust[mode]); */
move16();
st->log_en_adjust = add(mult(st->log_en_adjust, 29491),
shr(mult(shl(dtx_log_en_adjust[mode],5),3277),5));
/* Interpolate SID info */
int_fac = shl(add(1,st->since_last_sid), 10); /* Q10 */ move16();
int_fac = mult(int_fac, st->true_sid_period_inv); /* Q10 * Q15 -> Q10 */
/* Maximize to 1.0 in Q10 */
test();
if (sub(int_fac, 1024) > 0)
{
int_fac = 1024; move16();
}
int_fac = shl(int_fac, 4); /* Q10 -> Q14 */
L_log_en_int = L_mult(int_fac, st->log_en); /* Q14 * Q11->Q26 */ move32();
for(i = 0; i < M; i++)
{
lsp_int[i] = mult(int_fac, st->lsp[i]);/* Q14 * Q15 -> Q14 */ move16();
}
int_fac = sub(16384, int_fac); /* 1-k in Q14 */ move16();
/* (Q14 * Q11 -> Q26) + Q26 -> Q26 */
L_log_en_int = L_mac(L_log_en_int, int_fac, st->old_log_en);
for(i = 0; i < M; i++)
{
/* Q14 + (Q14 * Q15 -> Q14) -> Q14 */
lsp_int[i] = add(lsp_int[i], mult(int_fac, st->lsp_old[i])); move16();
lsp_int[i] = shl(lsp_int[i], 1); /* Q14 -> Q15 */ move16();
}
/* compute the amount of lsf variability */
lsf_variab_factor = sub(st->log_pg_mean,2457); /* -0.6 in Q12 */ move16();
/* *0.3 Q12*Q15 -> Q12 */
lsf_variab_factor = sub(4096, mult(lsf_variab_factor, 9830));
/* limit to values between 0..1 in Q12 */
test();
if (sub(lsf_variab_factor, 4096) > 0)
{
lsf_variab_factor = 4096; move16();
}
test();
if (lsf_variab_factor < 0)
{
lsf_variab_factor = 0; move16();
}
lsf_variab_factor = shl(lsf_variab_factor, 3); /* -> Q15 */ move16();
/* get index of vector to do variability with */
lsf_variab_index = pseudonoise(&st->L_pn_seed_rx, 3); move16();
/* convert to lsf */
Lsp_lsf(lsp_int, lsf_int, M);
/* apply lsf variability */
Copy(lsf_int, lsf_int_variab, M);
for(i = 0; i < M; i++)
{
move16();
lsf_int_variab[i] = add(lsf_int_variab[i],
mult(lsf_variab_factor,
st->lsf_hist_mean[i+lsf_variab_index*M]));
}
/* make sure that LSP's are ordered */
Reorder_lsf(lsf_int, LSF_GAP, M);
Reorder_lsf(lsf_int_variab, LSF_GAP, M);
/* copy lsf to speech decoders lsf state */
Copy(lsf_int, lsfState->past_lsf_q, M);
/* convert to lsp */
Lsf_lsp(lsf_int, lsp_int, M);
Lsf_lsp(lsf_int_variab, lsp_int_variab, M);
/* Compute acoeffs Q12 acoeff is used for level *
* normalization and postfilter, acoeff_variab is *
* used for synthesis filter *
* by doing this we make sure that the level *
* in high frequenncies does not jump up and down */
Lsp_Az(lsp_int, acoeff);
Lsp_Az(lsp_int_variab, acoeff_variab);
/* For use in postfilter */
Copy(acoeff, &A_t[0], M + 1);
Copy(acoeff, &A_t[M + 1], M + 1);
Copy(acoeff, &A_t[2 * (M + 1)], M + 1);
Copy(acoeff, &A_t[3 * (M + 1)], M + 1);
/* Compute reflection coefficients Q15 */
A_Refl(&acoeff[1], refl);
/* Compute prediction error in Q15 */
pred_err = MAX_16; /* 0.99997 in Q15 */ move16();
for (i = 0; i < M; i++)
{
pred_err = mult(pred_err, sub(MAX_16, mult(refl[i], refl[i])));
}
/* compute logarithm of prediction gain */
Log2(L_deposit_l(pred_err), &log_pg_e, &log_pg_m);
/* convert exponent and mantissa to Word16 Q12 */
log_pg = shl(sub(log_pg_e,15), 12); /* Q12 */ move16();
log_pg = shr(sub(0,add(log_pg, shr(log_pg_m, 15-12))), 1); move16();
st->log_pg_mean = add(mult(29491,st->log_pg_mean),
mult(3277, log_pg)); move16();
/* Compute interpolated log energy */
L_log_en_int = L_shr(L_log_en_int, 10); /* Q26 -> Q16 */ move32();
/* Add 4 in Q16 */
L_log_en_int = L_add(L_log_en_int, 4 * 65536L); move32();
/* subtract prediction gain */
L_log_en_int = L_sub(L_log_en_int, L_shl(L_deposit_l(log_pg), 4));move32();
/* adjust level to speech coder mode */
L_log_en_int = L_add(L_log_en_int,
L_shl(L_deposit_l(st->log_en_adjust), 5)); move32();
log_en_int_e = extract_h(L_log_en_int); move16();
move16();
log_en_int_m = extract_l(L_shr(L_sub(L_log_en_int,
L_deposit_h(log_en_int_e)), 1));
level = extract_l(Pow2(log_en_int_e, log_en_int_m)); /* Q4 */ move16();
for (i = 0; i < 4; i++)
{
/* Compute innovation vector */
build_CN_code(&st->L_pn_seed_rx, ex);
for (j = 0; j < L_SUBFR; j++)
{
ex[j] = mult(level, ex[j]); move16();
}
/* Synthesize */
Syn_filt(acoeff_variab, ex, &synth[i * L_SUBFR], L_SUBFR,
mem_syn, 1);
} /* next i */
/* reset codebook averaging variables */
averState->hangVar = 20; move16();
averState->hangCount = 0; move16();
test();
if (sub(new_state, DTX_MUTE) == 0)
{
/* mute comfort noise as it has been quite a long time since
* last SID update was performed */
tmp_int_length = st->since_last_sid; move16();
test();
if (sub(tmp_int_length, 32) > 0)
{
tmp_int_length = 32; move16();
}
/* safety guard against division by zero */
test();
if(tmp_int_length <= 0) {
tmp_int_length = 8; move16();
}
move16();
st->true_sid_period_inv = div_s(1 << 10, shl(tmp_int_length, 10));
st->since_last_sid = 0; move16();
Copy(st->lsp, st->lsp_old, M);
st->old_log_en = st->log_en; move16();
/* subtract 1/8 in Q11 i.e -6/8 dB */
st->log_en = sub(st->log_en, 256); move16();
}
/* reset interpolation length timer
* if data has been updated. */
test(); test(); test(); test();
if ((st->sid_frame != 0) &&
((st->valid_data != 0) ||
((st->valid_data == 0) && (st->dtxHangoverAdded) != 0)))
{
st->since_last_sid = 0; move16();
st->data_updated = 1; move16();
}
return 0;
}
/*
**************************************************************************
*
* Function : Decoder_amr
* Purpose : Speech decoder routine.
*
**************************************************************************
*/
int Decoder_amr (
Decoder_amrState *st, /* i/o : State variables */
enum Mode mode, /* i : AMR mode */
Word16 parm[], /* i : vector of synthesis parameters
(PRM_SIZE) */
enum RXFrameType frame_type, /* i : received frame type */
Word16 synth[], /* o : synthesis speech (L_FRAME) */
Word16 A_t[] /* o : decoded LP filter in 4 subframes
(AZ_SIZE) */
)
{
/* LPC coefficients */
Word16 *Az; /* Pointer on A_t */
/* LSPs */
Word16 lsp_new[M];
Word16 lsp_mid[M];
/* LSFs */
Word16 prev_lsf[M];
Word16 lsf_i[M];
/* Algebraic codevector */
Word16 code[L_SUBFR];
/* excitation */
Word16 excp[L_SUBFR];
Word16 exc_enhanced[L_SUBFR];
/* Scalars */
Word16 i, i_subfr;
Word16 T0, T0_frac, index, index_mr475 = 0;
Word16 gain_pit, gain_code, gain_code_mix, pit_sharp, pit_flag, pitch_fac;
Word16 t0_min, t0_max;
Word16 delta_frc_low, delta_frc_range;
Word16 tmp_shift;
Word16 temp;
Word32 L_temp;
Word16 flag4;
Word16 carefulFlag;
Word16 excEnergy;
Word16 subfrNr;
Word16 evenSubfr = 0;
Word16 bfi = 0; /* bad frame indication flag */
Word16 pdfi = 0; /* potential degraded bad frame flag */
enum DTXStateType newDTXState; /* SPEECH , DTX, DTX_MUTE */
/* find the new DTX state SPEECH OR DTX */
newDTXState = rx_dtx_handler(st->dtxDecoderState, frame_type);
move16 (); /* function result */
/* DTX actions */
test();
if (sub(newDTXState, SPEECH) != 0 )
{
Decoder_amr_reset (st, MRDTX);
dtx_dec(st->dtxDecoderState,
st->mem_syn,
st->lsfState,
st->pred_state,
st->Cb_gain_averState,
newDTXState,
mode,
parm, synth, A_t);
/* update average lsp */
Lsf_lsp(st->lsfState->past_lsf_q, st->lsp_old, M);
lsp_avg(st->lsp_avg_st, st->lsfState->past_lsf_q);
goto the_end;
}
/* SPEECH action state machine */
test (); test (); test ();
if ((sub(frame_type, RX_SPEECH_BAD) == 0) ||
(sub(frame_type, RX_NO_DATA) == 0) ||
(sub(frame_type, RX_ONSET) == 0))
{
bfi = 1; move16 ();
test(); test();
if ((sub(frame_type, RX_NO_DATA) == 0) ||
(sub(frame_type, RX_ONSET) == 0))
{
build_CN_param(&st->nodataSeed,
prmno[mode],
bitno[mode],
parm);
}
}
else if (sub(frame_type, RX_SPEECH_DEGRADED) == 0)
{
pdfi = 1; move16 ();
}
test();
if (bfi != 0)
{
st->state = add (st->state, 1);
}
else if (sub (st->state, 6) == 0)
{
st->state = 5; move16 ();
}
else
{
st->state = 0; move16 ();
}
test ();
if (sub (st->state, 6) > 0)
{
st->state = 6; move16 ();
}
/* If this frame is the first speech frame after CNI period, */
/* set the BFH state machine to an appropriate state depending */
/* on whether there was DTX muting before start of speech or not */
/* If there was DTX muting, the first speech frame is muted. */
/* If there was no DTX muting, the first speech frame is not */
/* muted. The BFH state machine starts from state 5, however, to */
/* keep the audible noise resulting from a SID frame which is */
/* erroneously interpreted as a good speech frame as small as */
/* possible (the decoder output in this case is quickly muted) */
test();
if (sub(st->dtxDecoderState->dtxGlobalState, DTX) == 0)
{
st->state = 5; move16 ();
st->prev_bf = 0; move16 ();
}
else if (test (), sub(st->dtxDecoderState->dtxGlobalState, DTX_MUTE) == 0)
{
st->state = 5; move16 ();
st->prev_bf = 1; move16 ();
}
/* save old LSFs for CB gain smoothing */
Copy (st->lsfState->past_lsf_q, prev_lsf, M);
/* decode LSF parameters and generate interpolated lpc coefficients
for the 4 subframes */
test ();
if (sub (mode, MR122) != 0)
{
D_plsf_3(st->lsfState, mode, bfi, parm, lsp_new);
fwc (); /* function worst case */
/* Advance synthesis parameters pointer */
parm += 3; move16 ();
Int_lpc_1to3(st->lsp_old, lsp_new, A_t);
}
else
{
D_plsf_5 (st->lsfState, bfi, parm, lsp_mid, lsp_new);
fwc (); /* function worst case */
/* Advance synthesis parameters pointer */
parm += 5; move16 ();
Int_lpc_1and3 (st->lsp_old, lsp_mid, lsp_new, A_t);
}
/* update the LSPs for the next frame */
for (i = 0; i < M; i++)
{
st->lsp_old[i] = lsp_new[i]; move16 ();
}
fwc (); /* function worst case */
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR *
* times *
* - decode the pitch delay *
* - decode algebraic code *
* - decode pitch and codebook gains *
* - find the excitation and compute synthesis speech *
*------------------------------------------------------------------------*/
/* pointer to interpolated LPC parameters */
Az = A_t; move16 ();
evenSubfr = 0; move16();
subfrNr = -1; move16();
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
subfrNr = add(subfrNr, 1);
evenSubfr = sub(1, evenSubfr);
/* flag for first and 3th subframe */
pit_flag = i_subfr; move16 ();
test();
if (sub (i_subfr, L_FRAME_BY2) == 0)
{
test(); test();
if (sub(mode, MR475) != 0 && sub(mode, MR515) != 0)
{
pit_flag = 0; move16 ();
}
}
/* pitch index */
index = *parm++; move16 ();
/*-------------------------------------------------------*
* - decode pitch lag and find adaptive codebook vector. *
*-------------------------------------------------------*/
test ();
if (sub(mode, MR122) != 0)
{
/* flag4 indicates encoding with 4 bit resolution; */
/* this is needed for mode MR475, MR515, MR59 and MR67 */
flag4 = 0; move16 ();
test (); test (); test (); test ();
if ((sub (mode, MR475) == 0) ||
(sub (mode, MR515) == 0) ||
(sub (mode, MR59) == 0) ||
(sub (mode, MR67) == 0) ) {
flag4 = 1; move16 ();
}
/*-------------------------------------------------------*
* - get ranges for the t0_min and t0_max *
* - only needed in delta decoding *
*-------------------------------------------------------*/
delta_frc_low = 5; move16();
delta_frc_range = 9; move16();
test ();
if ( sub(mode, MR795) == 0 )
{
delta_frc_low = 10; move16();
delta_frc_range = 19; move16();
}
t0_min = sub(st->old_T0, delta_frc_low);
test ();
if (sub(t0_min, PIT_MIN) < 0)
{
t0_min = PIT_MIN; move16();
}
t0_max = add(t0_min, delta_frc_range);
test ();
if (sub(t0_max, PIT_MAX) > 0)
{
t0_max = PIT_MAX; move16();
t0_min = sub(t0_max, delta_frc_range);
}
Dec_lag3 (index, t0_min, t0_max, pit_flag, st->old_T0,
&T0, &T0_frac, flag4);
st->T0_lagBuff = T0; move16 ();
test ();
if (bfi != 0)
{
test ();
if (sub (st->old_T0, PIT_MAX) < 0)
{ /* Graceful pitch */
st->old_T0 = add(st->old_T0, 1); /* degradation */
}
T0 = st->old_T0; move16 ();
T0_frac = 0; move16 ();
test (); test (); test (); test (); test ();
if ( st->inBackgroundNoise != 0 &&
sub(st->voicedHangover, 4) > 0 &&
((sub(mode, MR475) == 0 ) ||
(sub(mode, MR515) == 0 ) ||
(sub(mode, MR59) == 0) )
)
{
T0 = st->T0_lagBuff; move16 ();
}
}
fwc (); /* function worst case */
Pred_lt_3or6 (st->exc, T0, T0_frac, L_SUBFR, 1);
}
else
{
Dec_lag6 (index, PIT_MIN_MR122,
PIT_MAX, pit_flag, &T0, &T0_frac);
test (); test (); test ();
if ( bfi == 0 && (pit_flag == 0 || sub (index, 61) < 0))
{
}
else
{
st->T0_lagBuff = T0; move16 ();
T0 = st->old_T0; move16 ();
T0_frac = 0; move16 ();
}
fwc (); /* function worst case */
Pred_lt_3or6 (st->exc, T0, T0_frac, L_SUBFR, 0);
}
fwc (); /* function worst case */
/*-------------------------------------------------------*
* - (MR122 only: Decode pitch gain.) *
* - Decode innovative codebook. *
* - set pitch sharpening factor *
*-------------------------------------------------------*/
test (); test ();
if (sub (mode, MR475) == 0 || sub (mode, MR515) == 0)
{ /* MR475, MR515 */
index = *parm++; /* index of position */ move16 ();
i = *parm++; /* signs */ move16 ();
fwc (); /* function worst case */
decode_2i40_9bits (subfrNr, i, index, code);
fwc (); /* function worst case */
pit_sharp = shl (st->sharp, 1);
}
else if (sub (mode, MR59) == 0)
{ /* MR59 */
test ();
index = *parm++; /* index of position */ move16 ();
i = *parm++; /* signs */ move16 ();
fwc (); /* function worst case */
decode_2i40_11bits (i, index, code);
fwc (); /* function worst case */
pit_sharp = shl (st->sharp, 1);
}
else if (sub (mode, MR67) == 0)
{ /* MR67 */
test (); test ();
index = *parm++; /* index of position */ move16 ();
i = *parm++; /* signs */ move16 ();
fwc (); /* function worst case */
decode_3i40_14bits (i, index, code);
fwc (); /* function worst case */
pit_sharp = shl (st->sharp, 1);
}
else if (sub (mode, MR795) <= 0)
{ /* MR74, MR795 */
test (); test (); test ();
index = *parm++; /* index of position */ move16 ();
i = *parm++; /* signs */ move16 ();
fwc (); /* function worst case */
decode_4i40_17bits (i, index, code);
fwc (); /* function worst case */
pit_sharp = shl (st->sharp, 1);
}
else if (sub (mode, MR102) == 0)
{ /* MR102 */
test (); test (); test ();
fwc (); /* function worst case */
dec_8i40_31bits (parm, code);
parm += 7; move16 ();
fwc (); /* function worst case */
pit_sharp = shl (st->sharp, 1);
}
else
{ /* MR122 */
test (); test (); test ();
index = *parm++; move16 ();
test();
if (bfi != 0)
{
ec_gain_pitch (st->ec_gain_p_st, st->state, &gain_pit);
}
else
{
gain_pit = d_gain_pitch (mode, index); move16 ();
}
ec_gain_pitch_update (st->ec_gain_p_st, bfi, st->prev_bf,
&gain_pit);
fwc (); /* function worst case */
dec_10i40_35bits (parm, code);
parm += 10; move16 ();
fwc (); /* function worst case */
/* pit_sharp = gain_pit; */
/* if (pit_sharp > 1.0) pit_sharp = 1.0; */
pit_sharp = shl (gain_pit, 1);
}
/*-------------------------------------------------------*
* - Add the pitch contribution to code[]. *
*-------------------------------------------------------*/
for (i = T0; i < L_SUBFR; i++)
{
temp = mult (code[i - T0], pit_sharp);
code[i] = add (code[i], temp);
move16 ();
}
fwc (); /* function worst case */
/*------------------------------------------------------------*
* - Decode codebook gain (MR122) or both pitch *
* gain and codebook gain (all others) *
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
if (test(), sub (mode, MR475) == 0)
{
/* read and decode pitch and code gain */
test();
if (evenSubfr != 0)
{
index_mr475 = *parm++; move16 (); /* index of gain(s) */
}
test();
if (bfi == 0)
{
Dec_gain(st->pred_state, mode, index_mr475, code,
evenSubfr, &gain_pit, &gain_code);
}
else
{
ec_gain_pitch (st->ec_gain_p_st, st->state, &gain_pit);
ec_gain_code (st->ec_gain_c_st, st->pred_state, st->state,
&gain_code);
}
ec_gain_pitch_update (st->ec_gain_p_st, bfi, st->prev_bf,
&gain_pit);
ec_gain_code_update (st->ec_gain_c_st, bfi, st->prev_bf,
&gain_code);
fwc (); /* function worst case */
pit_sharp = gain_pit; move16 ();
test ();
if (sub (pit_sharp, SHARPMAX) > 0)
{
pit_sharp = SHARPMAX; move16 ();
}
}
else if (test(), test(), (sub (mode, MR74) <= 0) ||
(sub (mode, MR102) == 0))
{
/* read and decode pitch and code gain */
index = *parm++; move16 (); /* index of gain(s) */
test();
if (bfi == 0)
{
Dec_gain(st->pred_state, mode, index, code,
evenSubfr, &gain_pit, &gain_code);
}
else
{
ec_gain_pitch (st->ec_gain_p_st, st->state, &gain_pit);
ec_gain_code (st->ec_gain_c_st, st->pred_state, st->state,
&gain_code);
}
ec_gain_pitch_update (st->ec_gain_p_st, bfi, st->prev_bf,
&gain_pit);
ec_gain_code_update (st->ec_gain_c_st, bfi, st->prev_bf,
&gain_code);
fwc (); /* function worst case */
pit_sharp = gain_pit; move16 ();
test ();
if (sub (pit_sharp, SHARPMAX) > 0)
{
pit_sharp = SHARPMAX; move16 ();
}
if (sub (mode, MR102) == 0)
{
if (sub (st->old_T0, add(L_SUBFR, 5)) > 0)
{
pit_sharp = shr(pit_sharp, 2);
}
}
}
else
{
/* read and decode pitch gain */
index = *parm++; move16 (); /* index of gain(s) */
test ();
if (sub (mode, MR795) == 0)
{
/* decode pitch gain */
test();
if (bfi != 0)
{
ec_gain_pitch (st->ec_gain_p_st, st->state, &gain_pit);
}
else
{
gain_pit = d_gain_pitch (mode, index); move16 ();
}
ec_gain_pitch_update (st->ec_gain_p_st, bfi, st->prev_bf,
&gain_pit);
/* read and decode code gain */
index = *parm++; move16 ();
test();
if (bfi == 0)
{
d_gain_code (st->pred_state, mode, index, code, &gain_code);
}
else
{
ec_gain_code (st->ec_gain_c_st, st->pred_state, st->state,
&gain_code);
}
ec_gain_code_update (st->ec_gain_c_st, bfi, st->prev_bf,
&gain_code);
fwc (); /* function worst case */
pit_sharp = gain_pit; move16 ();
test ();
if (sub (pit_sharp, SHARPMAX) > 0)
{
pit_sharp = SHARPMAX; move16 ();
}
}
else
{ /* MR122 */
test();
if (bfi == 0)
{
d_gain_code (st->pred_state, mode, index, code, &gain_code);
}
else
{
ec_gain_code (st->ec_gain_c_st, st->pred_state, st->state,
&gain_code);
}
ec_gain_code_update (st->ec_gain_c_st, bfi, st->prev_bf,
&gain_code);
fwc (); /* function worst case */
pit_sharp = gain_pit; move16 ();
}
}
/* store pitch sharpening for next subframe */
/* (for modes which use the previous pitch gain for
pitch sharpening in the search phase) */
/* do not update sharpening in even subframes for MR475 */
test(); test();
if (sub(mode, MR475) != 0 || evenSubfr == 0)
{
st->sharp = gain_pit; move16 ();
test ();
if (sub (st->sharp, SHARPMAX) > 0)
{
st->sharp = SHARPMAX; move16 ();
}
}
pit_sharp = shl (pit_sharp, 1);
test ();
if (sub (pit_sharp, 16384) > 0)
{
for (i = 0; i < L_SUBFR; i++)
{
temp = mult (st->exc[i], pit_sharp);
L_temp = L_mult (temp, gain_pit);
test ();
if (sub(mode, MR122)==0)
{
L_temp = L_shr (L_temp, 1);
}
excp[i] = round (L_temp); move16 ();
}
}
/*-------------------------------------------------------*
* - Store list of LTP gains needed in the source *
* characteristic detector (SCD) *
*-------------------------------------------------------*/
test ();
if ( bfi == 0 )
{
for (i = 0; i < 8; i++)
{
st->ltpGainHistory[i] = st->ltpGainHistory[i+1]; move16 ();
}
st->ltpGainHistory[8] = gain_pit; move16 ();
}
/*-------------------------------------------------------*
* - Limit gain_pit if in background noise and BFI *
* for MR475, MR515, MR59 *
*-------------------------------------------------------*/
test (); test (); test (); test (); test (); test ();
if ( (st->prev_bf != 0 || bfi != 0) && st->inBackgroundNoise != 0 &&
((sub(mode, MR475) == 0) ||
(sub(mode, MR515) == 0) ||
(sub(mode, MR59) == 0))
)
{
test ();
if ( sub (gain_pit, 12288) > 0) /* if (gain_pit > 0.75) in Q14*/
gain_pit = add( shr( sub(gain_pit, 12288), 1 ), 12288 );
/* gain_pit = (gain_pit-0.75)/2.0 + 0.75; */
test ();
if ( sub (gain_pit, 14745) > 0) /* if (gain_pit > 0.90) in Q14*/
{
gain_pit = 14745; move16 ();
}
}
/*-------------------------------------------------------*
* Calculate CB mixed gain *
*-------------------------------------------------------*/
Int_lsf(prev_lsf, st->lsfState->past_lsf_q, i_subfr, lsf_i);
gain_code_mix = Cb_gain_average(
st->Cb_gain_averState, mode, gain_code,
lsf_i, st->lsp_avg_st->lsp_meanSave, bfi,
st->prev_bf, pdfi, st->prev_pdf,
st->inBackgroundNoise, st->voicedHangover); move16 ();
/* make sure that MR74, MR795, MR122 have original code_gain*/
test();
if ((sub(mode, MR67) > 0) && (sub(mode, MR102) != 0) )
/* MR74, MR795, MR122 */
{
gain_code_mix = gain_code; move16 ();
}
/*-------------------------------------------------------*
* - Find the total excitation. *
* - Find synthesis speech corresponding to st->exc[]. *
*-------------------------------------------------------*/
test ();
if (sub(mode, MR102) <= 0) /* MR475, MR515, MR59, MR67, MR74, MR795, MR102*/
{
pitch_fac = gain_pit; move16 ();
tmp_shift = 1; move16 ();
}
else /* MR122 */
{
pitch_fac = shr (gain_pit, 1); move16 ();
tmp_shift = 2; move16 ();
}
/* copy unscaled LTP excitation to exc_enhanced (used in phase
* dispersion below) and compute total excitation for LTP feedback
*/
for (i = 0; i < L_SUBFR; i++)
{
exc_enhanced[i] = st->exc[i]; move16 ();
/* st->exc[i] = gain_pit*st->exc[i] + gain_code*code[i]; */
L_temp = L_mult (st->exc[i], pitch_fac);
/* 12.2: Q0 * Q13 */
/* 7.4: Q0 * Q14 */
L_temp = L_mac (L_temp, code[i], gain_code);
/* 12.2: Q12 * Q1 */
/* 7.4: Q13 * Q1 */
L_temp = L_shl (L_temp, tmp_shift); /* Q16 */
st->exc[i] = round (L_temp); move16 ();
}
/*-------------------------------------------------------*
* - Adaptive phase dispersion *
*-------------------------------------------------------*/
ph_disp_release(st->ph_disp_st); /* free phase dispersion adaption */
test (); test (); test (); test (); test (); test ();
if ( ((sub(mode, MR475) == 0) ||
(sub(mode, MR515) == 0) ||
(sub(mode, MR59) == 0)) &&
sub(st->voicedHangover, 3) > 0 &&
st->inBackgroundNoise != 0 &&
bfi != 0 )
{
ph_disp_lock(st->ph_disp_st); /* Always Use full Phase Disp. */
} /* if error in bg noise */
/* apply phase dispersion to innovation (if enabled) and
compute total excitation for synthesis part */
ph_disp(st->ph_disp_st, mode,
exc_enhanced, gain_code_mix, gain_pit, code,
pitch_fac, tmp_shift);
/*-------------------------------------------------------*
* - The Excitation control module are active during BFI.*
* - Conceal drops in signal energy if in bg noise. *
*-------------------------------------------------------*/
L_temp = 0; move32 ();
for (i = 0; i < L_SUBFR; i++)
{
L_temp = L_mac (L_temp, exc_enhanced[i], exc_enhanced[i] );
}
L_temp = L_shr (L_temp, 1); /* excEnergy = sqrt(L_temp) in Q0 */
L_temp = sqrt_l_exp(L_temp, &temp); move32 (); /* function result */
L_temp = L_shr(L_temp, add( shr(temp, 1), 15));
L_temp = L_shr(L_temp, 2); /* To cope with 16-bit and */
excEnergy = extract_l(L_temp); /* scaling in ex_ctrl() */
test (); test (); test (); test (); test ();
test (); test (); test (); test (); test ();
if ( ((sub (mode, MR475) == 0) ||
(sub (mode, MR515) == 0) ||
(sub (mode, MR59) == 0)) &&
sub(st->voicedHangover, 5) > 0 &&
st->inBackgroundNoise != 0 &&
sub(st->state, 4) < 0 &&
( (pdfi != 0 && st->prev_pdf != 0) ||
bfi != 0 ||
st->prev_bf != 0) )
{
carefulFlag = 0; move32 ();
test (); test ();
if ( pdfi != 0 && bfi == 0 )
{
carefulFlag = 1; move16 ();
}
Ex_ctrl(exc_enhanced,
excEnergy,
st->excEnergyHist,
st->voicedHangover,
st->prev_bf,
carefulFlag);
}
test (); test (); test (); test ();
if ( st->inBackgroundNoise != 0 &&
( bfi != 0 || st->prev_bf != 0 ) &&
sub(st->state, 4) < 0 )
{
; /* do nothing! */
}
else
{
/* Update energy history for all modes */
for (i = 0; i < 8; i++)
{
st->excEnergyHist[i] = st->excEnergyHist[i+1]; move16 ();
}
st->excEnergyHist[8] = excEnergy; move16 ();
}
/*-------------------------------------------------------*
* Excitation control module end. *
*-------------------------------------------------------*/
fwc (); /* function worst case */
test ();
if (sub (pit_sharp, 16384) > 0)
{
for (i = 0; i < L_SUBFR; i++)
{
excp[i] = add (excp[i], exc_enhanced[i]);
move16 ();
}
agc2 (exc_enhanced, excp, L_SUBFR);
Overflow = 0; move16 ();
Syn_filt (Az, excp, &synth[i_subfr], L_SUBFR,
st->mem_syn, 0);
}
else
{
Overflow = 0; move16 ();
Syn_filt (Az, exc_enhanced, &synth[i_subfr], L_SUBFR,
st->mem_syn, 0);
}
test ();
if (Overflow != 0) /* Test for overflow */
{
for (i = 0; i < PIT_MAX + L_INTERPOL + L_SUBFR; i++)
{
st->old_exc[i] = shr(st->old_exc[i], 2); move16 ();
}
for (i = 0; i < L_SUBFR; i++)
{
exc_enhanced[i] = shr(exc_enhanced[i], 2); move16 ();
}
Syn_filt(Az, exc_enhanced, &synth[i_subfr], L_SUBFR, st->mem_syn, 1);
}
else
{
Copy(&synth[i_subfr+L_SUBFR-M], st->mem_syn, M);
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_SUBFR st->exc[] *
*--------------------------------------------------*/
Copy (&st->old_exc[L_SUBFR], &st->old_exc[0], PIT_MAX + L_INTERPOL);
fwc (); /* function worst case */
/* interpolated LPC parameters for next subframe */
Az += MP1; move16 ();
/* store T0 for next subframe */
st->old_T0 = T0; move16 ();
}
/*-------------------------------------------------------*
* Call the Source Characteristic Detector which updates *
* st->inBackgroundNoise and st->voicedHangover. *
*-------------------------------------------------------*/
move16 (); /* function result */
st->inBackgroundNoise = Bgn_scd(st->background_state,
&(st->ltpGainHistory[0]),
&(synth[0]),
&(st->voicedHangover) );
dtx_dec_activity_update(st->dtxDecoderState,
st->lsfState->past_lsf_q,
synth);
fwc (); /* function worst case */
/* store bfi for next subframe */
st->prev_bf = bfi; move16 ();
st->prev_pdf = pdfi; move16 ();
/*--------------------------------------------------*
* Calculate the LSF averages on the eight *
* previous frames *
*--------------------------------------------------*/
lsp_avg(st->lsp_avg_st, st->lsfState->past_lsf_q);
fwc (); /* function worst case */
the_end:
st->dtxDecoderState->dtxGlobalState = newDTXState; move16();
return 0;
}
void Decod_ld8kD(
Word16 parm[], /* (i) : vector of synthesis parameters
parm[0] = bad frame indicator (bfi) */
Word16 voicing, /* (i) : voicing decision from previous frame */
Word16 synth[], /* (o) : synthesis speech */
Word16 A_t[], /* (o) : decoded LP filter in 2 subframes */
Word16 *T0_first /* (o) : decoded pitch lag in first subframe */
)
{
Word16 *Az; /* Pointer on A_t */
Word16 lsp_new[M]; /* LSPs */
Word16 code[L_SUBFR]; /* ACELP codevector */
/* Scalars */
Word16 i, j, i_subfr;
Word16 T0, T0_frac, index;
Word16 bfi;
Word32 L_temp;
Word16 g_p, g_c; /* fixed and adaptive codebook gain */
Word16 bad_pitch; /* bad pitch indicator */
extern Flag Overflow;
/* Test bad frame indicator (bfi) */
bfi = *parm++;
/* Decode the LSPs */
D_lsp(parm, lsp_new, bfi);
parm += 2;
/* Interpolation of LPC for the 2 subframes */
Int_qlpc(lsp_old, lsp_new, A_t);
/* update the LSFs for the next frame */
Copy(lsp_new, lsp_old, M);
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR *
* times *
* - decode the pitch delay *
* - decode algebraic code *
* - decode pitch and codebook gains *
* - find the excitation and compute synthesis speech *
*------------------------------------------------------------------------*/
Az = A_t; /* pointer to interpolated LPC parameters */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
index = *parm++; /* pitch index */
if(i_subfr == 0)
{
if (sub(CODEC_MODE, 1) == 0) {
i = 0;
}
else if (sub(CODEC_MODE, 2) == 0) {
i = *parm++; /* get parity check result */
}
else {
fprintf(stderr, "CODEC mode invalid\n");
exit(-1);
}
bad_pitch = add(bfi, i);
if( bad_pitch == 0)
{
Dec_lag3D(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else /* Bad frame, or parity error */
{
T0 = old_T0;
T0_frac = 0;
old_T0 = add( old_T0, 1);
if( sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
*T0_first = T0; /* If first frame */
}
else /* second subframe */
{
if( bfi == 0)
{
Dec_lag3D(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else
{
T0 = old_T0;
T0_frac = 0;
old_T0 = add( old_T0, 1);
if( sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
}
/*-------------------------------------------------*
* - Find the adaptive codebook vector. *
*-------------------------------------------------*/
Pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);
/*-------------------------------------------------------*
* - Decode innovative codebook. *
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
if (sub(CODEC_MODE, 1) == 0) {
if(bfi != 0) { /* Bad frame */
parm[0] = Random() & (Word16)0x01ff; /* 9 bits random */
parm[1] = Random() & (Word16)0x0003; /* 2 bits random */
}
Decod_ACELP64(parm[1], parm[0], code);
}
else if (sub(CODEC_MODE, 2) == 0) {
if(bfi != 0) { /* Bad frame */
parm[0] = Random() & (Word16)0x1fff; /* 13 bits random */
parm[1] = Random() & (Word16)0x000f; /* 4 bits random */
}
Decod_ACELP(parm[1], parm[0], code);
}
else {
fprintf(stderr, "CODEC mode invalid\n");
exit(-1);
}
parm +=2;
j = shl(sharp, 1); /* From Q14 to Q15 */
if(sub(T0, L_SUBFR) <0 ) {
for (i = T0; i < L_SUBFR; i++) {
code[i] = add(code[i], mult(code[i-T0], j));
}
}
/*-------------------------------------------------*
* - Decode pitch and codebook gains. *
*-------------------------------------------------*/
index = *parm++; /* index of energy VQ */
if (sub(CODEC_MODE, 1) == 0) {
Dec_gain_6k(index, code, L_SUBFR, bfi, &gain_pitch, &gain_code, past_qua_en);
}
else if (sub(CODEC_MODE, 2) == 0) {
Dec_gain_8k(index, code, L_SUBFR, bfi, &gain_pitch, &gain_code, past_qua_en);
}
else {
fprintf(stderr, "CODEC mode invalid\n");
exit(-1);
}
/*-------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pitch *
*-------------------------------------------------------------*/
sharp = gain_pitch;
if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*-------------------------------------------------------*
* - Find the total excitation. *
* - Find synthesis speech corresponding to exc[]. *
*-------------------------------------------------------*/
if(bfi != 0) /* Bad frame */
{
if (voicing == 0 ) {
g_p = 0;
g_c = gain_code;
} else {
g_p = gain_pitch;
g_c = 0;
}
} else {
g_p = gain_pitch;
g_c = gain_code;
}
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = g_p*exc[i] + g_c*code[i]; */
/* exc[i] in Q0 g_p in Q14 */
/* code[i] in Q13 g_code in Q1 */
L_temp = L_mult(exc[i+i_subfr], g_p);
L_temp = L_mac(L_temp, code[i], g_c);
L_temp = L_shl(L_temp, 1);
exc[i+i_subfr] = round(L_temp);
}
/*-------------------------------------------------*
* Updates state machine for phase dispersion in *
* 6.4 kbps mode, if running in 8.0 kbps mode. *
*-------------------------------------------------*/
if (sub(CODEC_MODE, 2) == 0) {
Update64(gain_pitch, gain_code);
}
/* Test whether synthesis yields overflow or not */
Overflow = 0;
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0);
/* In case of overflow in the synthesis */
/* -> Scale down vector exc[] and redo synthesis */
if(Overflow != 0)
{
for(i=0; i<PIT_MAX+L_INTERPOL+L_FRAME; i++)
old_exc[i] = shr(old_exc[i], 2);
}
if (sub(CODEC_MODE, 1) == 0) {
Syn_filt64(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0,
gain_code, gain_pitch, code );
}
else {
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0);
}
/* Update of synthesis filter memory */
Copy(&synth[i_subfr+L_SUBFR-M], mem_syn, M);
Az += MP1; /* interpolated LPC parameters for next subframe */
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME exc[] *
*--------------------------------------------------*/
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
return;
}
